How To Register Sip In Bsnl Wings
The TATA SIP trunk will need a need a dedicated lan connectivity to your dialler,
so in dialer or pbx yous need ii Ethernet Port to connect to TATA network and a Dedicated network for phone registration and agent logins, every bit per the below pic.
eth1 is used to connect to TATA network, and eth0 is used to connect to customer lan network were ipphones , agents pc volition be connected. Once the above setup is set follow the below steps, if any support ping me in skype: striker24x7 STEP ane: Configure the TATA network ip to eth1 of dialer/pbx Either you lot tin utilise the below command to configure the tata ip in eth1 ifconfig eth1 10.0.seventy.18/30 OR edit the ifcfg-eth1 file by using below control *** Note : the ip accost ten.0.70.18 might be differ for each client. STEP ii: Configure the route in linux to attain tata network Goto and add below line Once file save restart the network by typing service network restart then type route -n to make sure road is added Pace 3: Asterisk sip settings. Goto 6 /etc/asteris/sip.conf and make below changes defaultexpiry=600 Pace four: Sip Carrier settings annals => 66810000:1234:66810000@10.0.lxx.2/66810000 [tatasip]
half dozen /etc/sysconfig/network-scripts/ifcfg-eth1
then enter
IPADDR=x.0.70.xviii
PREFIX=30
ONBOOT=YES
every bit SIP trunk network is on different subnet than the client IP, a static route needed
to attain the TATA SBC .
vi /etc/sysconfig/network-scripts/route-eth1
10.0.seventy.ii/32 via ten.0.70.71
progressinband=yes
For vicidial/goautodial yous can utilise the admin utility- Carrier settings
for plainly asterisk enter the below details in sip.conf.
blazon=friend
disallow=all
allow=alaw
allow=ulaw
allow=g729
host=10.0.70.2 ;this is tata SBC ip
dtmfmode=rfc2833
nat=no
canreinvite=no
context=tata
insecure=invite,port
STEP 5: Dialplan to dialout via tata trunk
For vicidial / goautodial yous can use the ADMIN-Carrier- Dialplan entry
For asterisk users you need to enter in extensions.conf under default contextFor vicidial/goautodial Dialers dialplan
exten => _7X.,i,AGI(agi://127.0.0.ane:4577/call_log)
exten => _7X.,due north,SipAddHeader(P-Preferred-Identity: <sip:66810000@ten.0.seventy.xviii>)
exten => _7X.,n,Progress()
exten => _7X.,n,Dial(SIP/0${EXTEN:1}@tatasip)
exten => _7X.,n,Hangup()
For asterisk/Freepbx Dialers dialplan
exten => _7X.,1,SipAddHeader(P-Preferred-Identity: <sip:66810000@10.0.70.eighteen>)
exten => _7X.,northward,Progress()
exten => _7X.,northward,Dial(SIP/0${EXTEN:one}@tatasip)
exten => _7X.,n,Hangup()
one time above entry done, practise a asterisk reload by typing asterisk -rx "reload" Enter the below dialplan later on the terminal line of extensions.conf (six /etc/asterisk/extensions.conf) For Vicidial/goautodial use the below dialplan in extensions.conf
STEP 6 : Dialplan to Receive entering calls form tatasip trunk
[tata]
exten => _X.,ane,Goto(s,1)
exten => s,1,Noop(Let united states wait deeper into the soul of the invite)
exten => south,n,Set(pseudodid=${SIP_HEADER(To)})
exten => south,n,Set(pseudodid=${Cut(pseudodid,@,1)})
exten => s,n,Fix(pseudodid=${CUT(pseudodid,:,2)})
exten => s,n,Goto(trunkinbound,${pseudodid},1)
Then in Vicidial GUI create DID's under Entering tab with your respective Tata DID no
for me its 66810000
For People using Freepbx/elastix/ or evidently asterisk who use from-pstn as inbound context use the below dialplan in extensions.conf
[tata]
exten => _X.,one,Goto(s,ane)
exten => s,1,Noop(Allow u.s. wait deeper into the soul of the invite)
exten => southward,northward,Set(pseudodid=${SIP_HEADER(To)})
exten => south,n,Fix(pseudodid=${CUT(pseudodid,@,1)})
exten => s,n,Prepare(pseudodid=${Cut(pseudodid,:,2)})
exten => s,n,Goto(from-pstn,${pseudodid},ane)
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Source: https://striker24x7.blogspot.com/2018/06/how-to-configure-tata-sip-voip-trunk-in.html
Posted by: kreidersonters.blogspot.com
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